webrtc/modules/audio_coding/main/test/EncodeToFileTest.h

80 lines
2.2 KiB
C++

/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef ENCODETOFILETEST_H
#define ENCODETOFILETEST_H
#include "ACMTest.h"
#include "audio_coding_module.h"
#include "typedefs.h"
#include "RTPFile.h"
#include "PCMFile.h"
#include <stdio.h>
using namespace webrtc;
// TestPacketization callback which writes the encoded payloads to file
class TestPacketization : public AudioPacketizationCallback
{
public:
TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
~TestPacketization();
virtual WebRtc_Word32 SendData(const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
const RTPFragmentationHeader* fragmentation);
private:
static void MakeRTPheader(WebRtc_UWord8* rtpHeader,
WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc);
RTPStream* _rtpStream;
WebRtc_Word32 _frequency;
WebRtc_Word16 _seqNo;
};
class Sender
{
public:
Sender();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
void Teardown();
void Run();
bool Add10MsData();
bool Process();
//for auto_test and logging
WebRtc_UWord8 testMode;
WebRtc_UWord8 codeId;
private:
AudioCodingModule* _acm;
PCMFile _pcmFile;
//WebRtc_Word16* _payloadData;
AudioFrame _audioFrame;
WebRtc_UWord16 _payloadSize;
WebRtc_UWord32 _timeStamp;
TestPacketization* _packetization;
};
// Test class
class EncodeToFileTest : public ACMTest
{
public:
EncodeToFileTest();
virtual void Perform(int fileType, int codeId, int* codePars, int testMode);
protected:
Sender _sender;
};
#endif